#37 in a series of articles about the technology behind Bang & Olufsen loudspeakers

In the last posting, I described a little experiment that we did where we could easily be accused of asking a stupid question and getting an obvious answer. This posting is a little different, although I’ll also describe a little experiment where we asked a much-less-stupid question and got an answer we weren’t looking for… However, before I describe the experiment itself, we’ll have to start with a little background information.

Let’s say that we wanted to build a “simple” two-way loudspeaker with a tweeter and a woofer in a box like the one shown in Figure 1.

Since we’re Bang & Olufsen, it is implied that this will be a fully active, DSP-based loudspeaker. Let’s also, for the sake of simplicity, say that we used loudspeaker drivers and other components with absolutely no linear distortion artefacts (I can dream, can’t I?). This means that we can apply any kind of filter we like to each of the two loudspeaker drivers to result in whatever response we want (in a given direction in a “free field” (a space with no reflecting surfaces, extending to infinity in all directions) – for more information on this, please read this posting and/or this posting).

Now, it’s fairly reasonable to assume that one portion of a good starting point for a loudspeaker’s acoustic response is to have a flat magnitude response when measured on-axis in a free field. This means that, if you have the loudspeaker in a space that has no reflections, and you are directly in front of it, no frequency will be louder than any other (within some limits, of course…). In fact, this is the way a lot of studio monitor loudspeakers are tuned (which makes sense, since many studio control rooms don’t have a lot of reflections as was discussed here).

The problem is that, if you do make a loudspeaker that is flat-on-axis, you’ll probably have problems in its power response (for a description of what a loudspeaker’s “power response” is, see here). This is dependent on a lot of things like the crossover frequency, the sizes and shapes of the drivers, the phase responses of the crossover filters, the shape of the cabinet, and other things. However, if we were to simplify, we could say that a two-way loudspeaker (say, with a 4th order Linkwitz-Riley crossover, just to pick one type…) that is flat on-axis, will have a dip in its power response at the crossover frequency. This is because, although the two drivers (the tweeter and the woofer) add together nicely in one location (on-axis, directly in front of the loudspeaker) they do not add together nicely anywhere else (because the distances to the two drivers from the listening position are probably not matched, particularly when you go vertical…).

So, one basic problem with building a “simple” two-way loudspeaker is that you have to choose to either have a flat magnitude response on-axis, or a smooth power response (without a dip) – but you can’t have both. (If you want to really dig into this, I’d recommend starting with J. Vanderkooy & S.P. Lipshitz, “Power Response of Loudspeakers with Noncoincident Drivers — The Influence of Crossover Design,” J. Audio Eng. Soc., Vol. 34, No. 4, pp. 236-244 (Apr. 1986).)

This basic problem raised a question in the head of one of our acoustical engineers, Gert Munch. He started to wonder how we could build a loudspeaker that could have a flat magnitude response on-axis and still have a smooth power response that didn’t suffer from a dip at the crossover. One possible solution is to build a loudspeaker with the desired on-axis response, and then somehow create an additional portion of the loudspeaker that could “fill up” the dip in the power response by sending energy into the room without it affecting the on-axis response.

One possible way to do this is to use an extra loudspeaker with a “dipole” characteristic – a two-sided loudspeaker where the same signal is sent to both sides, but in opposite polarity. This means that, when one side of the loudspeaker produces a high pressure, the other side produces a low. When you sit on one side of the loudspeaker or the other, then you hear a signal. However, if you sit “on edge” to the loudspeaker, you hear nothing, since the two sides of the loudspeaker cancel each other out (in theory).

So, Gert’s idea was to add a two-way dipole loudspeaker on the top of a normal two-way loudspeaker and to just use the dipole (which became known as a “shark fin” – shown in Figure 2)  to add energy around the crossover frequency of the “normal” loudspeaker. Since the edge of the dipole was facing forwards, the theory was that none of its sound would have an effect on the on-axis response of the two-way loudspeaker below it.

So, the question to answer was:

“Which sounds better:

• a loudspeaker tuned to be flat on-axis, but with no correction for its power response
• a loudspeaker that was tuned to have a smooth power response (in other words, with a “bump” around the crossover)
• a loudspeaker that was tuned to be flat on-axis and had a power response correction provided by the dipole

?”

Since each of the 6 loudspeaker drivers in our model loudspeaker was independently controllable, this was an easy comparison to do on-the-fly, so we tried it out.

Also, just to be certain, we tried two different orientations of the dipole, both of which had the “null” – the edge of the loudspeaker – facing the listening position. The first orientation is shown in Figures 2 and 3. The second orientation is shown in Figure 4.

So, a bunch of us sat in the listening room for a week or so, listening to the various possible tunings of the loudspeakers, in various positions in the room, with various types of music. We each had an opinion about the whether we liked one configuration more or less than another. However, one thing that was noticeable was that there was an effect not only on the timbral impression of the loudspeakers (the tone colour) – but also on the spatial presentation of the pair. As we switched between the different tunings, phantom images changed in width and distance, and envelopment (when it existed in the recording) also changed.

This was particularly noticeable when the loudspeakers were closer to a reflecting surface, as is shown in Figure 5.

The end result was a bunch of different conclusions:

1. The three different configurations of the loudspeaker (listed above) sounded different. (this was not a big surprise)
2. A two-way loudspeaker tuned to have a smooth power response and a loudspeaker with a flat on-axis response with a shark fin smoothing the power response sounded different – but mostly spatially.
3. Switching between those two loudspeaker configurations listed in Point #2 basically meant that we were switching between two different loudspeakers with the same power response, but different directivities (a fancy word meaning “directional behaviours”).
4. The dipole loudspeaker’s output was audible at the listening position because of the sidewall reflections in the case where the dipole was on top.
5. Switching the dipole on and off had a noticeable effect on the phantom image distance and width… When the dipole was turned on, the sidewall reflections were obviously louder, and the phantom images moved to a distance from the listening position roughly the same as the distance to the loudspeakers themselves. At the same time, the phantom image width increased and became “cloudy” or “fuzzy” and less precise.

All of those little conclusions could be folded into two big ones:

1. The original idea of using a dipole to fill in the power response of the loudspeaker doesn’t work well enough in all situations because we don’t know where the reflective surfaces around the loudspeaker will be in any given configuration.
2. Being able to control the directivity of the loudspeaker (specifically, how much energy is coming from the sidewall reflections) is a very interesting control parameter that we should look into…

So, this is where the concept of the shark fin died, but where the idea of “Beam Width Control” began.

Well… more accurately: it was the beginning of our thinking that being able to change the beam width (or the “directivity”, if you’re a geek) of the loudspeaker would be a useful “handle” in the hands of our customers. The idea that was born was something along the lines of the following: “If a customer could change the beam width of a loudspeaker, then (s)he could change it from being a narrow-directivity loudspeaker for people with one chair and no friends to a wide-directivity loudspeaker for people with a sofa and some friends to an omni-directivity loudspeaker for people throwing a party – all just by the flick of a switch…”

For more information on Beam Width Control:

What is “Beam Width Control”?

Beam Width Control – A Primer

#36 in a series of articles about the technology behind Bang & Olufsen loudspeakers

“In all affairs it’s a healthy thing now and then to hang a question mark on the things you have long taken for granted.”

-Bertrand Russell

If you were to get your hands on Harry Potter’s invisibility cloak and you were to walk into the acoustics department at Bang & Olufsen and eavesdrop on conversations, you’d sometimes be amazed at some of the questions we ask each other. That’s particularly true when we’re working on a new concept, because, if the concept is new, then it’s also new for us. The good thing is that all of my colleagues (and I) are ready and willing at any time to ask what might be considered to be a stupid (or at least a very basic) question.

One of those questions that we asked each other recently seemed like a basic one – why do we always put the tweeter on the top? It seems like there are very few loudspeakers that don’t do this (of course, there are exceptions – take the BeoLab Penta, for example, which has the tweeter in the middle). However, more often than not, when we (and most other people) make a loudspeaker, we put the loudspeaker drivers in ascending order of frequency – woofers on the bottom, tweeters on the top. Maybe this is because woofers are heavier, and if you stand a BeoLab 5 on its head, it will fall over – but that’s not really the question we were asking…

The REAL question we were asking ourselves at the time was: if we were to build a multi-way loudspeaker – let’s say a 3-way, with a woofer, a midrange and a tweeter, and if the crossovers were such that the bulk of the “interesting” information (say, the vocal range) was coming from the midrange, then why would we not put the midrange at ear height and put the tweeter between it and the woofer? For example, imagine we made BeoLab 20 without an Acoustic Lens on top, would it be better to arrange the drivers like the version on the left of Figure 1 or the version on the right? Which one would sound better?

After answering that question, there’s a second question would follow closely behind: how close together do the drivers with adjacent frequency bands (i.e. the woofer and the midrange or the tweeter and the midrange) have to be in order for them to sound like one thing? Of course, these two questions are inter-related. If your midrange and tweeter are so far apart that they sound like different sound sources, then you would probably be more interested in where the voice was coming from than where the cymbals were coming from…

Of course, step one in answering the second question could be to calculate/simulate the response of the loudspeaker, based on distance between drivers, the crossover frequency (and therefore the wavelengths of the frequency band we’re interested in), the slopes of the crossover filters, and so on. It would also be pretty easy to make a prototype model out of MDF, put the loudspeaker drivers in there, do the vertical directivity measurements of the system in the Cube, and see how well the theory matches reality.

However, the question we were really interested in was “but how would it sound?” – just to get a rough idea before going any further. And I have to stress here that we were really talking about getting a rough idea. What I’m about to describe to you is a little undertaking that we put together in a day – just to find out what would happen. This was not a big, scientifically-valid experiment using a large number of subjects and intensive statistics to evaluate the results. It was a couple of guys having a chat over coffee one morning when one of them said “I wonder what would happen if we put the midrange on top…” and then, off they went to a listening room to find out.

One thing we have learned in doing both “quick-n-dirty” listening comparisons and “real, scientifically valid listening tests” is that the placement of a loudspeaker in a room, has a huge effect on how the loudspeaker sounds. So, when we’re comparing two loudspeakers, we try to put them as close together as possible. So, we tried different versions of this. In the first, we took two pairs of loudspeakers, and put the left loudspeakers in each pair side-by-side, with one pair upside down and the other right-side up, as shown in Figure 2.

We then switched between the right-side up pair and the upside down pair, listening for a change in vertical position of the image. Note that we tried two arrangements of this – one where both right-side up loudspeakers were to the left of the upside-down loudspeakers. The other where the “right-side up” loudspeakers were the “outside” pair, as shown in Figure 3.

There are advantages and disadvantages of both of these arrangements – but in both cases, there is a lateral shift in the stereo image. When switching between pairs, either you get a left-right shift in image, or a change in width… It turned out that this change was more distracting than the vertical arrangement of the drivers, so we changed configuration to the one shown in Figure 4, below.

Now, instead of switching between loudspeakers, we pretended that one of them was a tweeter and other was a mid-woofer, and switched which was which, on the fly. Our “virtual” crossover was close-ish to the real crossover in the loudspeakers (our crossover was at 3.5 kHz, if you’re curious), so you could say that we were sort-of changing between using the upper tweeter + the lower woofer and using the upper woofer + lower tweeter, the “acoustical centres” of which are roughly at the same height. (remember – this was a quick-n-dirty experiment…)

After having listened to these three configurations of loudspeakers, we decided that the vertical arrangement of the drivers was not important with the vertical separation we were using.

This brought us to the second part of the question… If the tweeter and the midrange were further apart, would we have a preference? So, we kept our virtual crossover, using one loudspeaker as the “tweeter” and the other as the “mid-woofer”, and we moved the loudspeakers further apart, one example of which is shown in Figure 7. (One thing to note here is that when I say “further apart” I’m talking about the separation in the vertical angles of the sources – not necessarily their actual distance from each other. For example, if the loudspeakers were 1 m apart vertically, and you were level with one of the loudspeakers, but the listening position was a kilometre away, then the vertical angular separation (0.057 degrees) would be much smaller than if you were 1 m away (45 degrees)…)

The answer we arrived at at the end was that, when the vertical separation of the drivers gets extreme (perhaps I should use the word “silly” instead), we preferred the configuration where the “mid-woofer” was at ear-height. However, this was basically a choice of which version we disliked less (“preference” is a loaded word…). When the drivers get far enough apart, of course, they no longer “hang together” as a single device without some extra signal processing tricks.

So, we went to lunch having made the decision that, as long as the tweeters and the midranges are close enough to each other vertically, we really didn’t have a preference as to which was on top, so, if anyone asked, we would let the visual designers decide. (Remember that “close enough” is not only determined by this little experiment – it is also determined by the wavelengths of the crossover frequencies and whether or not there are more drivers to consider. For example, in the example in Figure 1, it might be that we don’t care about the relative placement of the tweeter and midrange in isolation – but perhaps putting the tweeter between the midrange and woofer will make them too far apart to have a nice vertical directivity behaviour across the lower crossover…)

Addendum

This isn’t the first time we asked ourselves such a question. Although I was not yet working at B&O at the time, my colleagues tell me that, back in the days when they were developing the BeoLab 3500 and the BeoLab 7-1 (both of which are “stereo” loudspeakers – essentially two multi-way loudspeakers in a single device) , they questioned the driver arrangement as well. Should the tweeters or the midranges / woofers be on the outside? You can see in the photos of the 3500 on this page that they decided to put the lower-frequency bands wider because the overall impression with stereo signals was wider than if the tweeters were on the outside.

#35 in a series of articles about the technology behind Bang & Olufsen loudspeakers

Introduction:
Why do I need to compensate for my room?

Take a look at Figure 1. You’ll see a pair of headphones (BeoPlay H6‘s, if you’re curious…) sitting under a lamp that is lighting them directly. (That lamp is the only light source in the room. I can’t prove it, so you’ll have to trust me on this one…) You can see the headphones because the light is shining on them, right? Well… sort of.

What happens if we put something between the lamp and the headphones? Take a look at Figure 2, which was taken with the same camera, the same lens, the same shutter speed, the same F-stop, and the same ISO (in other words, I’m not playing any tricks on you – I promise…).

Notice that you can still see the headphones, even though there is no direct light shining on them. This probably does not come as a surprise, since there is a mirror next to them – so there is enough light bouncing off the mirror to reflect enough light back to the headphones so that we can still see them. In fact, there’s enough light from the mirror that we can see the shadow caused by the reflected lamp (which is also visible in Figure 1, if you’re paying attention…).

If you don’t believe me, look around the room you’re sitting in right now. You can probably see everything in it – even the things that do not have light shining directly on them (for example, the wall behind an open door, or the floor beneath your feet if you lift them a little…)

Exactly the same is true for sound. Let’s turn the lamp into a loudspeaker and the headphones on the floor into you, in the listening position and send a “click” sound (what we geeks call an “impulse”) out of the loudspeaker. What arrives at the listening position? This is illustrated in Figure 3, which is what we call an “impulse response” – how a room responds to an impulse (a click coming from a loudspeaker).

The top plot in Figure 1 shows the signal that is sent to the input of the loudspeaker. The bottom plot is the signal at the input of the microphone placed at the listening room. If we zoom in on the bottom plot, the result is Figure 4. This makes it much easier to see the direct sound and the reflections as separate components.

If we zoom in even further to the beginning of the plot in Figure 4, we can see individual reflections in the room’s response, as is shown in Figure 5.

Let’s take the total impulse response and separate the direct sound from the rest. This is shown in Figure 6.

We can then calculate the magnitude responses of the two separate components individually to see what their relative contributions are – shown in Figure 7.

Now, before you get carried away, I have to say up-front that this plot is a little misleading for many reasons – but I’ll only mention two…

The first is that it shows that the direct sound is quieter than the reflected sound in almost all frequency bands, but as you can see in Figure 6, the reflected energy is never actually louder than the direct sound. However, the reflected energy lasts for much longer than the direct sound, which is why the analysis “sees” it as containing more energy – but you don’t hear the decay in the room’s response at the same time when you play a click out of the loudspeaker. Then again, you usually don’t listen to a click – you listen to music, so you’re listening to the end of the room decay on the music that happened a second ago while you’re listening to the middle of the decay on the music that happened a half-second ago while you’re listening to the direct sound of the music that happened just now… So, at any given time, if you’re playing music (assuming that this music was constantly loud – like Metallica, for example…), you’re hearing a lot of energy from the room smearing music from the recent past, compared to the amount of energy in the direct sound which is the most recent thing to come out of the loudspeaker.

The second is in the apparent magnitude response of the direct sound. It appears from the red curve in Figure 7 that this loudspeaker has a response that lacks low frequency energy. This is not actually true – the loudspeaker that I used for this measurement actually has a flat on-axis magnitude response within about 1 dB from well below 20 Hz to well above 20 kHz. However, in order to see that actual response of the loudspeaker, I would have to use a much longer slice of time than the little red spike shown in Figure 6. In other words, the weirdness in the magnitude response is an artefact of the time-slicing of the impulse response. The details of this are complicated, so I won’t bother explaining it in this article – you’ll just have to trust me when I say that that isn’t really the actual response of the loudspeaker in free space…

The “punch line” for all of this is that the room has a significant influence on the perceived sound of the loudspeaker (something I talked about in more detail in this article). The more reflective the surfaces in the room, the more influence it has on the sound. (Also, the more omnidirectional the loudspeaker, the more energy it sends in more directions in the room, which also will mean that the room has more influence on the total sound at the listening position… but there’s more information about that in the article on Beam Width Control.)

So, if the room has a significant influence on the sound of the loudspeaker at the listening position, then it’s smart to want to do something about it. In a best case (and very generally speaking…), we would want to measure the effects that the room has on the overall sound of the loudspeaker and “undo” them. The problem is that we can’t actually undo them without changing the room itself. However, we can make some compensation for some aspects of the effects of the room. For example, one of the obvious things in the blue curve in Figure 7 is that the listening room I did the measurement in  has a nasty resonance in the low end (specifically, it’s at about 57 Hz which is the second axial mode for the depth of the room which is about 6 m). It would certainly help the overall sound of the loudspeaker to put in a notch filter at that frequency – in a best case, we should measure the phase response of the room’s resonance and insert a filter that has the opposite phase response. But that’s just the beginning with one mode – there are lots more things to fix…

A short history

Almost all Bang & Olufsen loudspeakers have a switch that allows you to change its magnitude response to compensate for the position of the loudspeaker in the room. This is typically called a Free/Wall/Corner switch, since it’s designed to offset the changes to the timbre of the loudspeaker caused by the closest boundaries. There’s a whole article about this effect and how we make a filter to compensate for it at this link.

In 2002, Bang & Olufsen took this a step further when it introduced the BeoLab 5 which included ABC – Automatic Bass Calibration. This was a system that uses a microphone to measure the effects of the listening room’s acoustical behaviour on the sound of the loudspeaker, and then creates a filter that compensates for those effects in the low frequency band. As a simple example, if your room tends to increase the apparent bass level, then the BeoLab 5’s reduce their bass level by the same amount. This system works very well, but it has some drawbacks. Specifically, ABC is designed to improve the response of the loudspeaker averaged over all locations in the room. However this follows the philosophy first stated Spock said in Star Trek II: The Wrath of Kahn when he said “the needs of the many outweigh the needs of the few, or the one.” In other words, in order to make the averaged response of the loudspeaker better in all locations in the room, it could be that the response at one location in the room (say, the “sweet spot” for example…) gets worse (whatever that might mean…). This philosophy behind ABC makes sense in BeoLab 5, since it is designed as a loudspeaker that has a wide horizontal directivity – meaning it is designed as a loudspeaker for “social” listening, not as a loudspeaker for someone with one chair and no friends… Therefore an improved average room response would “win” in importance over an improved sweet spot.

Active Room Compensation

We are currently working on a taking this concept to a new level with Active Room Compensation. Using an external microphone, we can measure the effects of the room’s acoustical behaviour in different zones in the room and subsequently optimise compensation filters for different situations. For example, in order to duplicate the behaviour of BeoLab 5’s ABC, we just need to use the microphone to measure a number of widely-space locations around the room, thus giving us a total average for the space. However, if we want to create a room compensation filter for a single location – the sweet spot, for example – then we can restrict the locations of the microphone measurements to that area within the room. If we want to have a compensation filter that is pretty good for the whole room, but has emphasis on the sweet spot, we just have to make more measurements in the sweet spot than in the rest of the room. The weighting of importance of different locations in the room can be determined by the number of microphone measurements we do in each location. Of course, this isn’s as simple a procedure as pressing one button, as in ABC on the BeoLab 5, but it has the advantage in the ability to create a compensation filter for a specific location instead of for the whole listening space.

As part of this work, we are developing a new concept in acoustical room compensation: multichannel processing. This means that the loudspeakers not only “see” each other as having an effect on the room – but they help each other to control the room’s acoustical influence. So, if you play music in the left loudspeaker only, then some sound will also come out of the right loudspeaker. This is because both the left and right loudspeakers are working together to control the room (which is being “activated” by sound only from the left loudspeaker.